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[alsa-devel] [PATCH] Patch Dreamcast AICA driver to handle Maple bus support
by Adrian McMenamin 10 Sep '07

10 Sep '07

This patches the AICA sound driver for the Dreamcast to handle the well known flakiness of the Dreamcast's G2 bus. This is dependent on getting Maple bus support (see http://lkml.org/lkml/2007/9/9/70) into the kernel. Signed-off by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 7397865..a42c58f 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -35,6 +35,7 @@ #include <linux/timer.h> #include <linux/delay.h> #include <linux/workqueue.h> +#include <linux/maple.h> #include <sound/driver.h> #include <sound/core.h> #include <sound/control.h> @@ -43,7 +44,7 @@ #include <sound/info.h> #include <asm/io.h> #include <asm/dma.h> -#include <asm/dreamcast/sysasic.h> +#include <asm/mach/sysasic.h> #include "aica.h" MODULE_AUTHOR("Adrian McMenamin <adrian(a)mcmen.demon.co.uk>"); @@ -218,6 +219,12 @@ static int aica_dma_transfer(int channels, int buffer_size, period_offset = dreamcastcard->clicks; period_offset %= (AICA_PERIOD_NUMBER / channels); runtime = substream->runtime; + /* If maple dma is running, wait for it to finish */ + do{ cpu_relax();} + while (!maple_dma_done()); + /* Turn off the maple dma now */ + ctrl_outl(0, MAPLE_ENABLE); + local_irq_disable(); for (q = 0; q < channels; q++) { err = dma_xfer(AICA_DMA_CHANNEL, (unsigned long) (runtime->dma_area + @@ -232,6 +239,9 @@ static int aica_dma_transfer(int channels, int buffer_size, break; dma_wait_for_completion(AICA_DMA_CHANNEL); } + /* restore maple dma */ + local_irq_enable(); + maplebus_init_hardware(); return err; }

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[alsa-devel] [PATCH] ad1848/cs4231: replace commented out debug code with snd-printd{, d}
by Rene Herman 10 Sep '07

10 Sep '07

Hi Takashi. While I'm at it another "while I'm there" -- replace commented out debug code with snd-printd{,d}. (I saw Krysztof was busy tweaking the sleeping granularity in these functions, so I'm leaving the code alone after this). Rene. diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c index 161da11..ace2a9b 100644 --- a/sound/isa/ad1848/ad1848_lib.c +++ b/sound/isa/ad1848/ad1848_lib.c @@ -211,13 +211,14 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) /* end of cleanup sequence */ for (timeout = 12000; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) udelay(100); -#if 0 - printk("(1) timeout = %i\n", timeout); -#endif + + snd_printdd("(1) timeout = %d\n", timeout); + #ifdef CONFIG_SND_DEBUG if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) snd_printk(KERN_WARNING "mce_down [0x%lx] - auto calibration time out (0)\n", AD1848P(chip, REGSEL)); #endif + chip->mce_bit &= ~AD1848_MCE; timeout = inb(AD1848P(chip, REGSEL)); outb(chip->mce_bit | (timeout & 0x1f), AD1848P(chip, REGSEL)); @@ -234,9 +235,9 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) * which at the slowest possible rate of 5.5125 kHz means 907 us. */ msleep(1); -#if 0 - printk("(2) jiffies = %li\n", jiffies); -#endif + + snd_printdd("(2) jiffies = %lu\n", jiffies); + time = msecs_to_jiffies(250); while (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) { spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -247,9 +248,9 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) time = schedule_timeout_interruptible(time); spin_lock_irqsave(&chip->reg_lock, flags); } -#if 0 - printk("(3) jiffies = %li\n", jiffies); -#endif + + snd_printdd("(3) jiffies = %lu\n", jiffies); + time = msecs_to_jiffies(100); while (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) { spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -261,10 +262,9 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) spin_lock_irqsave(&chip->reg_lock, flags); } spin_unlock_irqrestore(&chip->reg_lock, flags); -#if 0 - printk("(4) jiffies = %li\n", jiffies); - snd_printk("mce_down - exit = 0x%x\n", inb(AD1848P(chip, REGSEL))); -#endif + + snd_printdd("(4) jiffies = %lu\n", jiffies); + snd_printd("mce_down - exit = 0x%x\n", inb(AD1848P(chip, REGSEL))); } static unsigned int snd_ad1848_get_count(unsigned char format, diff --git a/sound/isa/cs423x/cs4231_lib.c b/sound/isa/cs423x/cs4231_lib.c index 9e8e0f1..d3a7b59 100644 --- a/sound/isa/cs423x/cs4231_lib.c +++ b/sound/isa/cs423x/cs4231_lib.c @@ -326,9 +326,7 @@ void snd_cs4231_mce_down(struct snd_cs4231 *chip) int timeout; snd_cs4231_busy_wait(chip); -#if 0 - printk("(1) timeout = %i\n", timeout); -#endif + #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("mce_down [0x%lx] - auto calibration time out (0)\n", (long)CS4231P(REGSEL)); @@ -352,9 +350,9 @@ void snd_cs4231_mce_down(struct snd_cs4231 *chip) * which at the slowest possible rate of 5.5125 kHz means 907 us. */ msleep(1); -#if 0 - printk("(2) jiffies = %li\n", jiffies); -#endif + + snd_printdd("(1) jiffies = %lu\n", jiffies); + /* in 10 ms increments, check condition, up to 250 ms */ timeout = 25; while (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) { @@ -364,9 +362,9 @@ void snd_cs4231_mce_down(struct snd_cs4231 *chip) } msleep(10); } -#if 0 - printk("(3) jiffies = %li\n", jiffies); -#endif + + snd_printdd("(2) jiffies = %lu\n", jiffies); + /* in 10 ms increments, check condition, up to 100 ms */ timeout = 10; while (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) { @@ -376,10 +374,9 @@ void snd_cs4231_mce_down(struct snd_cs4231 *chip) } msleep(10); } -#if 0 - printk("(4) jiffies = %li\n", jiffies); - snd_printk("mce_down - exit = 0x%x\n", cs4231_inb(chip, CS4231P(REGSEL))); -#endif + + snd_printdd("(3) jiffies = %lu\n", jiffies); + snd_printd("mce_down - exit = 0x%x\n", cs4231_inb(chip, CS4231P(REGSEL))); } static unsigned int snd_cs4231_get_count(unsigned char format, unsigned int size)

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[alsa-devel] [PATCH] ad1848: fix AD1848P macro
by Rene Herman 10 Sep '07

10 Sep '07

Hi. Consistent variable naming is a good thing, but let's be a little less sneaky about enforcing it... ;-/ Rene. diff --git a/include/sound/ad1848.h b/include/sound/ad1848.h index b2c3f00..ca0b6c3 100644 --- a/include/sound/ad1848.h +++ b/include/sound/ad1848.h @@ -27,7 +27,7 @@ /* IO ports */ -#define AD1848P( codec, x ) ( (chip) -> port + c_d_c_AD1848##x ) +#define AD1848P( chip, x ) ( (chip) -> port + c_d_c_AD1848##x ) #define c_d_c_AD1848REGSEL 0 #define c_d_c_AD1848REG 1

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[alsa-devel] [PATCH] hda: BIOS changing subsystem id
by Matthew Ranostay 10 Sep '07

10 Sep '07

Some laptop BIOS change the subsystem id for STAC9205 cards if the microphone isn't toggled on/off in the settings. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r f2c43802b1aa pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Aug 21 15:20:26 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Thu Aug 30 15:24:59 2007 -0400 @@ -824,6 +824,8 @@ static struct snd_pci_quirk stac9205_cfg "Dell Precision", STAC_9205_M43xx), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f9, "Dell Precision", STAC_9205_DELL_M43), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021b, + "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fa, "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fe,

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[alsa-devel] Tascam driver on arm
by ST 10 Sep '07

10 Sep '07

Hi I just tried to build the tascam sound driver SND_USB_USX2Y on arm. But this is not allowed in the Kconfig of the module. Is there a reason for this? Thanks ST

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[alsa-devel] Unable to bypass alsa plugins and write directly to hw:0, 0
by Pharaoh . 10 Sep '07

10 Sep '07

My sound card supports hardware upsampling i.e. I do not need to use the software up sampling done by plugins like dmix etc. For testing hw up sampling, I tried to play an 8K sample rate wav file as follows, but some how it is being upsampled, I think I am missing something. # aplay -Dhw:0,0 8k.wav -v Playing WAVE '8k.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Stereo Warning: rate is not accurate (requested = 8000Hz, got = 44100Hz) please, try the plug plugin Hardware PCM card 0 'OMAP850 EAC' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22052 period_size : 5513 period_time : 125011 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 5513 xfer_align : 5513 start_threshold : 22052 stop_threshold : 22052 silence_threshold: 0 silence_size : 0 boundary : 1445199872 I have attached my alsa.conf. -pharaoh.

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[alsa-devel] Fwd: Re: RME HDSPe Info/Question
by ricman 10 Sep '07

10 Sep '07

Posting reply from kind responder providing additional info... --- Colin Stewart Campbell <colin.campbell(a)comhem.se> wrote: > From: Colin Stewart Campbell <colin.campbell(a)comhem.se> > Subject: Re: [alsa-devel] RME HDSPe Info/Question > Date: Mon, 10 Sep 2007 14:23:28 +0200 > To: ricman <the_ricman(a)yahoo.com> > > Well, except for that the new PCIe cards do all the floating-point > conversion in hardware and present 32-bit float instead of 32-bit > integer data on the bus. > If the alsa driver handles this, then ok. If not then you'll have to > > wait. Those running OSX on Intel machines with PCI slots already got > > hit by this format issue. > > Colin > ____________________________________________________________________________________ Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC

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[alsa-devel] [PATCH] ad1848: replace HZ calculus with msecs_to_jiffies()
by Rene Herman 10 Sep '07

10 Sep '07

Hi Takashi. If I'm not mistaken, any (new) use of HZ these days is considered a bug so while I'm there... Signed-off-by: Rene Herman <rene.herman(a)gmail.com> Rene. diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c index dcd4435..161da11 100644 --- a/sound/isa/ad1848/ad1848_lib.c +++ b/sound/isa/ad1848/ad1848_lib.c @@ -237,7 +237,7 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) #if 0 printk("(2) jiffies = %li\n", jiffies); #endif - time = HZ / 4; + time = msecs_to_jiffies(250); while (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) { spin_unlock_irqrestore(&chip->reg_lock, flags); if (time <= 0) { @@ -250,7 +250,7 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) #if 0 printk("(3) jiffies = %li\n", jiffies); #endif - time = HZ / 10; + time = msecs_to_jiffies(100); while (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) { spin_unlock_irqrestore(&chip->reg_lock, flags); if (time <= 0) {

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[alsa-devel] [PATCH] ad1848 and cs4231 busy loop replacement
by Krzysztof Helt 10 Sep '07

10 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch replaces busy loop with msleep and removes incorrect debug message. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This should fix Rene's problem with the timeout message he got in his new Aztech driver. The problem is that the driver wait for busy bit to go high but it may not happen if the auto-calibration is not set. The result is incorrect debug message (that timeout happened). It is also possible (in theory) to get this message in the ad1848-lib on a slow CPU. The patch just wait a period around the period needed for the auto-calibration. If there is no auto-calibration set, the driver just loses few miliseconds. This may happen only during driver start as both drivers set auto-calibration. If the auto-calibration is set we give CPU more time to do something else then move to next waiting loop (for the auto-calibration bit to get cleared). The msleep delay is calculated as rounded down time of waiting number of cycles (from ad1848k and cs4231a specs) with the highest frequency (48kHz). Rene, please test this patch on the cs4231 codec. diff -urp linux-2.6.22/sound/isa/ad1848/ad1848_lib.c linux-2.6.23/sound/isa/ad1848/ad1848_lib.c --- linux-2.6.22/sound/isa/ad1848/ad1848_lib.c 2007-09-08 23:53:42.958067597 +0200 +++ linux-2.6.23/sound/isa/ad1848/ad1848_lib.c 2007-09-08 23:57:31.255077502 +0200 @@ -229,17 +229,12 @@ static void snd_ad1848_mce_down(struct s spin_unlock_irqrestore(&chip->reg_lock, flags); return; } + /* calibration process */ + msleep(7); + + snd_printd("(2) jiffies = %li\n", jiffies); - for (timeout = 500; timeout > 0 && (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) == 0; timeout--); - if ((snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) == 0) { - snd_printd("mce_down - auto calibration time out (1)\n"); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return; - } -#if 0 - printk("(2) timeout = %i, jiffies = %li\n", timeout, jiffies); -#endif time = HZ / 4; while (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) { spin_unlock_irqrestore(&chip->reg_lock, flags); diff -urp linux-2.6.22/sound/isa/cs423x/cs4231_lib.c linux-2.6.23/sound/isa/cs423x/cs4231_lib.c --- linux-2.6.22/sound/isa/cs423x/cs4231_lib.c 2007-09-08 23:53:29.005272473 +0200 +++ linux-2.6.23/sound/isa/cs423x/cs4231_lib.c 2007-09-08 23:56:52.824887490 +0200 @@ -334,19 +334,12 @@ void snd_cs4231_mce_down(struct snd_cs42 !(chip->hardware & (CS4231_HW_CS4231_MASK | CS4231_HW_CS4232_MASK))) { return; } - snd_cs4231_busy_wait(chip); /* calibration process */ + msleep(2); + + snd_printd("(2) jiffies = %li\n", jiffies); - for (timeout = 500; timeout > 0 && (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0; timeout--) - udelay(10); - if ((snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0) { - snd_printd("cs4231_mce_down - auto calibration time out (1)\n"); - return; - } -#if 0 - printk("(2) timeout = %i, jiffies = %li\n", timeout, jiffies); -#endif /* in 10 ms increments, check condition, up to 250 ms */ timeout = 25; while (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) {

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[alsa-devel] alsaseq 0.1 Python module
by Patricio Paez 10 Sep '07

10 Sep '07

Hello all, This was posted in the Python announce list, but it could be of interest here: "The first version of alsaseq, bindings to the ALSA sequencer has been released. alsaseq is a Python module that allows to interact with ALSA sequencer clients. It can create an ALSA client, connect to other clients, send and receive ALSA events immediately or at a scheduled time using a sequencer queue. It provides a subset of the ALSA sequencer capabilities in a simplified model. It is implemented in C language and licensed under the Gnu GPL license version 2 or later. Home http://pp.com.mx/python/alsaseq Download http://pp.com.mx/python/alsaseq/alsaseq-0.1.tar.gz " It is an early and ongoing project, suggestions and comments are welcome. Regards, Patricio Páez <pp at pp.com.mx> Original post: http://mail.python.org/pipermail/python-announce-list/2007-September/006102…

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